In many types of apparatus, analog signals are converted into digital signals by analog-to-digital converters known as A/D's. After the digital signals are processed, the results are converted to an output signal in analog form by a digital-to-analog converter known as a D/A and a low pass filter. The digital signals are formed at a sampling rate 1/t.sub.0 that is at least twice the highest analog signal frequency of interest. Whereas this method of recovering the output analog signal works in a satisfactory manner for most applications, it introduces objectionable distortion in applications where extreme accuracy is desired. This arises from the fact that in accordance with the Nyquist theorem, the low pass filter will produce the desired analog output signal with precision only if the signal samples applied to it have zero width, i.e., if they are a series of Dirac delta functions. However, most practicable D/A converters output a stepped wave having the height of each sample until the next sample is received t.sub.0 seconds later. This is equivalent to convolving the narrow pulses required by the Nyquist theorem with a rectangular pulse. Fourrier analysis indicates that such a pulse has a Sin .omega.t/.omega. distribution of frequency components so that the convolution causes an output signal 0(.omega.) in the frequency domain as indicated by the following expression, wherein H is the desired signal and t.sub.0 is the width of a step or the time between samples: ##EQU1## The factor (Sin .omega.t.sub.0 /2)/.omega. causes the distortion referred to.